Sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!

Asterisk 11.21.2
Freepbx 13.0.79

I have three chan_sip extension. It was working until I reboot (VPS) server

When I tried to reboot the phone, It will register and stay online for approx 30 seconds
Then it will switch to unreachable. The error message I get is sip_poke_noanswer: Peer ‘XXX’ is now UNREACHABLE!

Under Advance Setting (sip.conf) I tried the following setting in the parenthesis but same result
qualify=yes (no | 3000)
qualifyfreq=60 (15 | 20 | 90| 120)

While status is Unreachable, I can still make outgoing calls.

Debug report
<— SIP read from UDP:81.36.xxx.xxx:5086 —>
REGISTER sip:pbx3.domain.net SIP/2.0
Via: SIP/2.0/UDP 10.1.11.198:5086;branch=z9hG4bK-a0872809;rport
From: “313” sip:[email protected];tag=9d03ac11eada9b73o0
To: “313” sip:[email protected]
Call-ID: [email protected]
CSeq: 5363 REGISTER
Max-Forwards: 70
Authorization: Digest username=“313”,realm=“asterisk”,nonce=“16d536d0”,uri=“sip:pbx3.domain.net”,algorithm=MD5,response="780a4583616103d1dde12dc1b6d06886"
Contact: “313” sip:[email protected]:5086;expires=3600
User-Agent: Cisco/SPA303-7.5.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
— (13 headers 0 lines) —
Sending to 81.36.xxx.xxx:5086 (NAT)
– Registered SIP ‘313’ at 81.36.xxx.xxx:5086
Reliably Transmitting (NAT) to 81.36.xxx.xxx:5086:
OPTIONS sip:[email protected]:5086 SIP/2.0
Via: SIP/2.0/UDP 162.221.xxx.xxx:5060;branch=z9hG4bK5dbf820a;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as14f39802
To: sip:[email protected]:5086
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.79(11.21.2)
Date: Mon, 14 Mar 2016 10:34:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Appreciate any help

Same exact issue!
Did you ever get anywhere with this?

same issue here, did you ever resolv @beachcomp or @rsarceno

Tried to change the nat settings in advanced asterisk sip?
Is your VPS fixed ip?

Same issue. Can plug these phones into a different network and they work perfectly. On this network they become unreachable after ~30 seconds.

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